diff options
Diffstat (limited to '')
| -rw-r--r-- | media/libopus/include/opus.h | 981 |
1 files changed, 981 insertions, 0 deletions
diff --git a/media/libopus/include/opus.h b/media/libopus/include/opus.h new file mode 100644 index 0000000000..0c69c627d9 --- /dev/null +++ b/media/libopus/include/opus.h @@ -0,0 +1,981 @@ +/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited + Written by Jean-Marc Valin and Koen Vos */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/** + * @file opus.h + * @brief Opus reference implementation API + */ + +#ifndef OPUS_H +#define OPUS_H + +#include "opus_types.h" +#include "opus_defines.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** + * @mainpage Opus + * + * The Opus codec is designed for interactive speech and audio transmission over the Internet. + * It is designed by the IETF Codec Working Group and incorporates technology from + * Skype's SILK codec and Xiph.Org's CELT codec. + * + * The Opus codec is designed to handle a wide range of interactive audio applications, + * including Voice over IP, videoconferencing, in-game chat, and even remote live music + * performances. It can scale from low bit-rate narrowband speech to very high quality + * stereo music. Its main features are: + + * @li Sampling rates from 8 to 48 kHz + * @li Bit-rates from 6 kb/s to 510 kb/s + * @li Support for both constant bit-rate (CBR) and variable bit-rate (VBR) + * @li Audio bandwidth from narrowband to full-band + * @li Support for speech and music + * @li Support for mono and stereo + * @li Support for multichannel (up to 255 channels) + * @li Frame sizes from 2.5 ms to 60 ms + * @li Good loss robustness and packet loss concealment (PLC) + * @li Floating point and fixed-point implementation + * + * Documentation sections: + * @li @ref opus_encoder + * @li @ref opus_decoder + * @li @ref opus_repacketizer + * @li @ref opus_multistream + * @li @ref opus_libinfo + * @li @ref opus_custom + */ + +/** @defgroup opus_encoder Opus Encoder + * @{ + * + * @brief This page describes the process and functions used to encode Opus. + * + * Since Opus is a stateful codec, the encoding process starts with creating an encoder + * state. This can be done with: + * + * @code + * int error; + * OpusEncoder *enc; + * enc = opus_encoder_create(Fs, channels, application, &error); + * @endcode + * + * From this point, @c enc can be used for encoding an audio stream. An encoder state + * @b must @b not be used for more than one stream at the same time. Similarly, the encoder + * state @b must @b not be re-initialized for each frame. + * + * While opus_encoder_create() allocates memory for the state, it's also possible + * to initialize pre-allocated memory: + * + * @code + * int size; + * int error; + * OpusEncoder *enc; + * size = opus_encoder_get_size(channels); + * enc = malloc(size); + * error = opus_encoder_init(enc, Fs, channels, application); + * @endcode + * + * where opus_encoder_get_size() returns the required size for the encoder state. Note that + * future versions of this code may change the size, so no assuptions should be made about it. + * + * The encoder state is always continuous in memory and only a shallow copy is sufficient + * to copy it (e.g. memcpy()) + * + * It is possible to change some of the encoder's settings using the opus_encoder_ctl() + * interface. All these settings already default to the recommended value, so they should + * only be changed when necessary. The most common settings one may want to change are: + * + * @code + * opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate)); + * opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity)); + * opus_encoder_ctl(enc, OPUS_SET_SIGNAL(signal_type)); + * @endcode + * + * where + * + * @arg bitrate is in bits per second (b/s) + * @arg complexity is a value from 1 to 10, where 1 is the lowest complexity and 10 is the highest + * @arg signal_type is either OPUS_AUTO (default), OPUS_SIGNAL_VOICE, or OPUS_SIGNAL_MUSIC + * + * See @ref opus_encoderctls and @ref opus_genericctls for a complete list of parameters that can be set or queried. Most parameters can be set or changed at any time during a stream. + * + * To encode a frame, opus_encode() or opus_encode_float() must be called with exactly one frame (2.5, 5, 10, 20, 40 or 60 ms) of audio data: + * @code + * len = opus_encode(enc, audio_frame, frame_size, packet, max_packet); + * @endcode + * + * where + * <ul> + * <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li> + * <li>frame_size is the duration of the frame in samples (per channel)</li> + * <li>packet is the byte array to which the compressed data is written</li> + * <li>max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended). + * Do not use max_packet to control VBR target bitrate, instead use the #OPUS_SET_BITRATE CTL.</li> + * </ul> + * + * opus_encode() and opus_encode_float() return the number of bytes actually written to the packet. + * The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value + * is 2 bytes or less, then the packet does not need to be transmitted (DTX). + * + * Once the encoder state if no longer needed, it can be destroyed with + * + * @code + * opus_encoder_destroy(enc); + * @endcode + * + * If the encoder was created with opus_encoder_init() rather than opus_encoder_create(), + * then no action is required aside from potentially freeing the memory that was manually + * allocated for it (calling free(enc) for the example above) + * + */ + +/** Opus encoder state. + * This contains the complete state of an Opus encoder. + * It is position independent and can be freely copied. + * @see opus_encoder_create,opus_encoder_init + */ +typedef struct OpusEncoder OpusEncoder; + +/** Gets the size of an <code>OpusEncoder</code> structure. + * @param[in] channels <tt>int</tt>: Number of channels. + * This must be 1 or 2. + * @returns The size in bytes. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels); + +/** + */ + +/** Allocates and initializes an encoder state. + * There are three coding modes: + * + * @ref OPUS_APPLICATION_VOIP gives best quality at a given bitrate for voice + * signals. It enhances the input signal by high-pass filtering and + * emphasizing formants and harmonics. Optionally it includes in-band + * forward error correction to protect against packet loss. Use this + * mode for typical VoIP applications. Because of the enhancement, + * even at high bitrates the output may sound different from the input. + * + * @ref OPUS_APPLICATION_AUDIO gives best quality at a given bitrate for most + * non-voice signals like music. Use this mode for music and mixed + * (music/voice) content, broadcast, and applications requiring less + * than 15 ms of coding delay. + * + * @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY configures low-delay mode that + * disables the speech-optimized mode in exchange for slightly reduced delay. + * This mode can only be set on an newly initialized or freshly reset encoder + * because it changes the codec delay. + * + * This is useful when the caller knows that the speech-optimized modes will not be needed (use with caution). + * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz) + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal + * @param [in] application <tt>int</tt>: Coding mode (one of @ref OPUS_APPLICATION_VOIP, @ref OPUS_APPLICATION_AUDIO, or @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY) + * @param [out] error <tt>int*</tt>: @ref opus_errorcodes + * @note Regardless of the sampling rate and number channels selected, the Opus encoder + * can switch to a lower audio bandwidth or number of channels if the bitrate + * selected is too low. This also means that it is safe to always use 48 kHz stereo input + * and let the encoder optimize the encoding. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusEncoder *opus_encoder_create( + opus_int32 Fs, + int channels, + int application, + int *error +); + +/** Initializes a previously allocated encoder state + * The memory pointed to by st must be at least the size returned by opus_encoder_get_size(). + * This is intended for applications which use their own allocator instead of malloc. + * @see opus_encoder_create(),opus_encoder_get_size() + * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL. + * @param [in] st <tt>OpusEncoder*</tt>: Encoder state + * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz) + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal + * @param [in] application <tt>int</tt>: Coding mode (one of OPUS_APPLICATION_VOIP, OPUS_APPLICATION_AUDIO, or OPUS_APPLICATION_RESTRICTED_LOWDELAY) + * @retval #OPUS_OK Success or @ref opus_errorcodes + */ +OPUS_EXPORT int opus_encoder_init( + OpusEncoder *st, + opus_int32 Fs, + int channels, + int application +) OPUS_ARG_NONNULL(1); + +/** Encodes an Opus frame. + * @param [in] st <tt>OpusEncoder*</tt>: Encoder state + * @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16) + * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the + * input signal. + * This must be an Opus frame size for + * the encoder's sampling rate. + * For example, at 48 kHz the permitted + * values are 120, 240, 480, 960, 1920, + * and 2880. + * Passing in a duration of less than + * 10 ms (480 samples at 48 kHz) will + * prevent the encoder from using the LPC + * or hybrid modes. + * @param [out] data <tt>unsigned char*</tt>: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated + * memory for the output + * payload. This may be + * used to impose an upper limit on + * the instant bitrate, but should + * not be used as the only bitrate + * control. Use #OPUS_SET_BITRATE to + * control the bitrate. + * @returns The length of the encoded packet (in bytes) on success or a + * negative error code (see @ref opus_errorcodes) on failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode( + OpusEncoder *st, + const opus_int16 *pcm, + int frame_size, + unsigned char *data, + opus_int32 max_data_bytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + +/** Encodes an Opus frame from floating point input. + * @param [in] st <tt>OpusEncoder*</tt>: Encoder state + * @param [in] pcm <tt>float*</tt>: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0. + * Samples with a range beyond +/-1.0 are supported but will + * be clipped by decoders using the integer API and should + * only be used if it is known that the far end supports + * extended dynamic range. + * length is frame_size*channels*sizeof(float) + * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the + * input signal. + * This must be an Opus frame size for + * the encoder's sampling rate. + * For example, at 48 kHz the permitted + * values are 120, 240, 480, 960, 1920, + * and 2880. + * Passing in a duration of less than + * 10 ms (480 samples at 48 kHz) will + * prevent the encoder from using the LPC + * or hybrid modes. + * @param [out] data <tt>unsigned char*</tt>: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated + * memory for the output + * payload. This may be + * used to impose an upper limit on + * the instant bitrate, but should + * not be used as the only bitrate + * control. Use #OPUS_SET_BITRATE to + * control the bitrate. + * @returns The length of the encoded packet (in bytes) on success or a + * negative error code (see @ref opus_errorcodes) on failure. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode_float( + OpusEncoder *st, + const float *pcm, + int frame_size, + unsigned char *data, + opus_int32 max_data_bytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + +/** Frees an <code>OpusEncoder</code> allocated by opus_encoder_create(). + * @param[in] st <tt>OpusEncoder*</tt>: State to be freed. + */ +OPUS_EXPORT void opus_encoder_destroy(OpusEncoder *st); + +/** Perform a CTL function on an Opus encoder. + * + * Generally the request and subsequent arguments are generated + * by a convenience macro. + * @param st <tt>OpusEncoder*</tt>: Encoder state. + * @param request This and all remaining parameters should be replaced by one + * of the convenience macros in @ref opus_genericctls or + * @ref opus_encoderctls. + * @see opus_genericctls + * @see opus_encoderctls + */ +OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NONNULL(1); +/**@}*/ + +/** @defgroup opus_decoder Opus Decoder + * @{ + * + * @brief This page describes the process and functions used to decode Opus. + * + * The decoding process also starts with creating a decoder + * state. This can be done with: + * @code + * int error; + * OpusDecoder *dec; + * dec = opus_decoder_create(Fs, channels, &error); + * @endcode + * where + * @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000 + * @li channels is the number of channels (1 or 2) + * @li error will hold the error code in case of failure (or #OPUS_OK on success) + * @li the return value is a newly created decoder state to be used for decoding + * + * While opus_decoder_create() allocates memory for the state, it's also possible + * to initialize pre-allocated memory: + * @code + * int size; + * int error; + * OpusDecoder *dec; + * size = opus_decoder_get_size(channels); + * dec = malloc(size); + * error = opus_decoder_init(dec, Fs, channels); + * @endcode + * where opus_decoder_get_size() returns the required size for the decoder state. Note that + * future versions of this code may change the size, so no assuptions should be made about it. + * + * The decoder state is always continuous in memory and only a shallow copy is sufficient + * to copy it (e.g. memcpy()) + * + * To decode a frame, opus_decode() or opus_decode_float() must be called with a packet of compressed audio data: + * @code + * frame_size = opus_decode(dec, packet, len, decoded, max_size, 0); + * @endcode + * where + * + * @li packet is the byte array containing the compressed data + * @li len is the exact number of bytes contained in the packet + * @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float()) + * @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array + * + * opus_decode() and opus_decode_float() return the number of samples (per channel) decoded from the packet. + * If that value is negative, then an error has occurred. This can occur if the packet is corrupted or if the audio + * buffer is too small to hold the decoded audio. + * + * Opus is a stateful codec with overlapping blocks and as a result Opus + * packets are not coded independently of each other. Packets must be + * passed into the decoder serially and in the correct order for a correct + * decode. Lost packets can be replaced with loss concealment by calling + * the decoder with a null pointer and zero length for the missing packet. + * + * A single codec state may only be accessed from a single thread at + * a time and any required locking must be performed by the caller. Separate + * streams must be decoded with separate decoder states and can be decoded + * in parallel unless the library was compiled with NONTHREADSAFE_PSEUDOSTACK + * defined. + * + */ + +/** Opus decoder state. + * This contains the complete state of an Opus decoder. + * It is position independent and can be freely copied. + * @see opus_decoder_create,opus_decoder_init + */ +typedef struct OpusDecoder OpusDecoder; + +/** Gets the size of an <code>OpusDecoder</code> structure. + * @param [in] channels <tt>int</tt>: Number of channels. + * This must be 1 or 2. + * @returns The size in bytes. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_size(int channels); + +/** Allocates and initializes a decoder state. + * @param [in] Fs <tt>opus_int32</tt>: Sample rate to decode at (Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode + * @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes + * + * Internally Opus stores data at 48000 Hz, so that should be the default + * value for Fs. However, the decoder can efficiently decode to buffers + * at 8, 12, 16, and 24 kHz so if for some reason the caller cannot use + * data at the full sample rate, or knows the compressed data doesn't + * use the full frequency range, it can request decoding at a reduced + * rate. Likewise, the decoder is capable of filling in either mono or + * interleaved stereo pcm buffers, at the caller's request. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusDecoder *opus_decoder_create( + opus_int32 Fs, + int channels, + int *error +); + +/** Initializes a previously allocated decoder state. + * The state must be at least the size returned by opus_decoder_get_size(). + * This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size + * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL. + * @param [in] st <tt>OpusDecoder*</tt>: Decoder state. + * @param [in] Fs <tt>opus_int32</tt>: Sampling rate to decode to (Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode + * @retval #OPUS_OK Success or @ref opus_errorcodes + */ +OPUS_EXPORT int opus_decoder_init( + OpusDecoder *st, + opus_int32 Fs, + int channels +) OPUS_ARG_NONNULL(1); + +/** Decode an Opus packet. + * @param [in] st <tt>OpusDecoder*</tt>: Decoder state + * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss + * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload* + * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length + * is frame_size*channels*sizeof(opus_int16) + * @param [in] frame_size Number of samples per channel of available space in \a pcm. + * If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will + * not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1), + * then frame_size needs to be exactly the duration of audio that is missing, otherwise the + * decoder will not be in the optimal state to decode the next incoming packet. For the PLC and + * FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms. + * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be + * decoded. If no such data is available, the frame is decoded as if it were lost. + * @returns Number of decoded samples or @ref opus_errorcodes + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode( + OpusDecoder *st, + const unsigned char *data, + opus_int32 len, + opus_int16 *pcm, + int frame_size, + int decode_fec +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Decode an Opus packet with floating point output. + * @param [in] st <tt>OpusDecoder*</tt>: Decoder state + * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss + * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload + * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length + * is frame_size*channels*sizeof(float) + * @param [in] frame_size Number of samples per channel of available space in \a pcm. + * If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will + * not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1), + * then frame_size needs to be exactly the duration of audio that is missing, otherwise the + * decoder will not be in the optimal state to decode the next incoming packet. For the PLC and + * FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms. + * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be + * decoded. If no such data is available the frame is decoded as if it were lost. + * @returns Number of decoded samples or @ref opus_errorcodes + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode_float( + OpusDecoder *st, + const unsigned char *data, + opus_int32 len, + float *pcm, + int frame_size, + int decode_fec +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Perform a CTL function on an Opus decoder. + * + * Generally the request and subsequent arguments are generated + * by a convenience macro. + * @param st <tt>OpusDecoder*</tt>: Decoder state. + * @param request This and all remaining parameters should be replaced by one + * of the convenience macros in @ref opus_genericctls or + * @ref opus_decoderctls. + * @see opus_genericctls + * @see opus_decoderctls + */ +OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NONNULL(1); + +/** Frees an <code>OpusDecoder</code> allocated by opus_decoder_create(). + * @param[in] st <tt>OpusDecoder*</tt>: State to be freed. + */ +OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st); + +/** Parse an opus packet into one or more frames. + * Opus_decode will perform this operation internally so most applications do + * not need to use this function. + * This function does not copy the frames, the returned pointers are pointers into + * the input packet. + * @param [in] data <tt>char*</tt>: Opus packet to be parsed + * @param [in] len <tt>opus_int32</tt>: size of data + * @param [out] out_toc <tt>char*</tt>: TOC pointer + * @param [out] frames <tt>char*[48]</tt> encapsulated frames + * @param [out] size <tt>opus_int16[48]</tt> sizes of the encapsulated frames + * @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes) + * @returns number of frames + */ +OPUS_EXPORT int opus_packet_parse( + const unsigned char *data, + opus_int32 len, + unsigned char *out_toc, + const unsigned char *frames[48], + opus_int16 size[48], + int *payload_offset +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5); + +/** Gets the bandwidth of an Opus packet. + * @param [in] data <tt>char*</tt>: Opus packet + * @retval OPUS_BANDWIDTH_NARROWBAND Narrowband (4kHz bandpass) + * @retval OPUS_BANDWIDTH_MEDIUMBAND Mediumband (6kHz bandpass) + * @retval OPUS_BANDWIDTH_WIDEBAND Wideband (8kHz bandpass) + * @retval OPUS_BANDWIDTH_SUPERWIDEBAND Superwideband (12kHz bandpass) + * @retval OPUS_BANDWIDTH_FULLBAND Fullband (20kHz bandpass) + * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_bandwidth(const unsigned char *data) OPUS_ARG_NONNULL(1); + +/** Gets the number of samples per frame from an Opus packet. + * @param [in] data <tt>char*</tt>: Opus packet. + * This must contain at least one byte of + * data. + * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz. + * This must be a multiple of 400, or + * inaccurate results will be returned. + * @returns Number of samples per frame. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs) OPUS_ARG_NONNULL(1); + +/** Gets the number of channels from an Opus packet. + * @param [in] data <tt>char*</tt>: Opus packet + * @returns Number of channels + * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsigned char *data) OPUS_ARG_NONNULL(1); + +/** Gets the number of frames in an Opus packet. + * @param [in] packet <tt>char*</tt>: Opus packet + * @param [in] len <tt>opus_int32</tt>: Length of packet + * @returns Number of frames + * @retval OPUS_BAD_ARG Insufficient data was passed to the function + * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1); + +/** Gets the number of samples of an Opus packet. + * @param [in] packet <tt>char*</tt>: Opus packet + * @param [in] len <tt>opus_int32</tt>: Length of packet + * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz. + * This must be a multiple of 400, or + * inaccurate results will be returned. + * @returns Number of samples + * @retval OPUS_BAD_ARG Insufficient data was passed to the function + * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, opus_int32 Fs) OPUS_ARG_NONNULL(1); + +/** Gets the number of samples of an Opus packet. + * @param [in] dec <tt>OpusDecoder*</tt>: Decoder state + * @param [in] packet <tt>char*</tt>: Opus packet + * @param [in] len <tt>opus_int32</tt>: Length of packet + * @returns Number of samples + * @retval OPUS_BAD_ARG Insufficient data was passed to the function + * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2); + +/** Applies soft-clipping to bring a float signal within the [-1,1] range. If + * the signal is already in that range, nothing is done. If there are values + * outside of [-1,1], then the signal is clipped as smoothly as possible to + * both fit in the range and avoid creating excessive distortion in the + * process. + * @param [in,out] pcm <tt>float*</tt>: Input PCM and modified PCM + * @param [in] frame_size <tt>int</tt> Number of samples per channel to process + * @param [in] channels <tt>int</tt>: Number of channels + * @param [in,out] softclip_mem <tt>float*</tt>: State memory for the soft clipping process (one float per channel, initialized to zero) + */ +OPUS_EXPORT void opus_pcm_soft_clip(float *pcm, int frame_size, int channels, float *softclip_mem); + + +/**@}*/ + +/** @defgroup opus_repacketizer Repacketizer + * @{ + * + * The repacketizer can be used to merge multiple Opus packets into a single + * packet or alternatively to split Opus packets that have previously been + * merged. Splitting valid Opus packets is always guaranteed to succeed, + * whereas merging valid packets only succeeds if all frames have the same + * mode, bandwidth, and frame size, and when the total duration of the merged + * packet is no more than 120 ms. The 120 ms limit comes from the + * specification and limits decoder memory requirements at a point where + * framing overhead becomes negligible. + * + * The repacketizer currently only operates on elementary Opus + * streams. It will not manipualte multistream packets successfully, except in + * the degenerate case where they consist of data from a single stream. + * + * The repacketizing process starts with creating a repacketizer state, either + * by calling opus_repacketizer_create() or by allocating the memory yourself, + * e.g., + * @code + * OpusRepacketizer *rp; + * rp = (OpusRepacketizer*)malloc(opus_repacketizer_get_size()); + * if (rp != NULL) + * opus_repacketizer_init(rp); + * @endcode + * + * Then the application should submit packets with opus_repacketizer_cat(), + * extract new packets with opus_repacketizer_out() or + * opus_repacketizer_out_range(), and then reset the state for the next set of + * input packets via opus_repacketizer_init(). + * + * For example, to split a sequence of packets into individual frames: + * @code + * unsigned char *data; + * int len; + * while (get_next_packet(&data, &len)) + * { + * unsigned char out[1276]; + * opus_int32 out_len; + * int nb_frames; + * int err; + * int i; + * err = opus_repacketizer_cat(rp, data, len); + * if (err != OPUS_OK) + * { + * release_packet(data); + * return err; + * } + * nb_frames = opus_repacketizer_get_nb_frames(rp); + * for (i = 0; i < nb_frames; i++) + * { + * out_len = opus_repacketizer_out_range(rp, i, i+1, out, sizeof(out)); + * if (out_len < 0) + * { + * release_packet(data); + * return (int)out_len; + * } + * output_next_packet(out, out_len); + * } + * opus_repacketizer_init(rp); + * release_packet(data); + * } + * @endcode + * + * Alternatively, to combine a sequence of frames into packets that each + * contain up to <code>TARGET_DURATION_MS</code> milliseconds of data: + * @code + * // The maximum number of packets with duration TARGET_DURATION_MS occurs + * // when the frame size is 2.5 ms, for a total of (TARGET_DURATION_MS*2/5) + * // packets. + * unsigned char *data[(TARGET_DURATION_MS*2/5)+1]; + * opus_int32 len[(TARGET_DURATION_MS*2/5)+1]; + * int nb_packets; + * unsigned char out[1277*(TARGET_DURATION_MS*2/2)]; + * opus_int32 out_len; + * int prev_toc; + * nb_packets = 0; + * while (get_next_packet(data+nb_packets, len+nb_packets)) + * { + * int nb_frames; + * int err; + * nb_frames = opus_packet_get_nb_frames(data[nb_packets], len[nb_packets]); + * if (nb_frames < 1) + * { + * release_packets(data, nb_packets+1); + * return nb_frames; + * } + * nb_frames += opus_repacketizer_get_nb_frames(rp); + * // If adding the next packet would exceed our target, or it has an + * // incompatible TOC sequence, output the packets we already have before + * // submitting it. + * // N.B., The nb_packets > 0 check ensures we've submitted at least one + * // packet since the last call to opus_repacketizer_init(). Otherwise a + * // single packet longer than TARGET_DURATION_MS would cause us to try to + * // output an (invalid) empty packet. It also ensures that prev_toc has + * // been set to a valid value. Additionally, len[nb_packets] > 0 is + * // guaranteed by the call to opus_packet_get_nb_frames() above, so the + * // reference to data[nb_packets][0] should be valid. + * if (nb_packets > 0 && ( + * ((prev_toc & 0xFC) != (data[nb_packets][0] & 0xFC)) || + * opus_packet_get_samples_per_frame(data[nb_packets], 48000)*nb_frames > + * TARGET_DURATION_MS*48)) + * { + * out_len = opus_repacketizer_out(rp, out, sizeof(out)); + * if (out_len < 0) + * { + * release_packets(data, nb_packets+1); + * return (int)out_len; + * } + * output_next_packet(out, out_len); + * opus_repacketizer_init(rp); + * release_packets(data, nb_packets); + * data[0] = data[nb_packets]; + * len[0] = len[nb_packets]; + * nb_packets = 0; + * } + * err = opus_repacketizer_cat(rp, data[nb_packets], len[nb_packets]); + * if (err != OPUS_OK) + * { + * release_packets(data, nb_packets+1); + * return err; + * } + * prev_toc = data[nb_packets][0]; + * nb_packets++; + * } + * // Output the final, partial packet. + * if (nb_packets > 0) + * { + * out_len = opus_repacketizer_out(rp, out, sizeof(out)); + * release_packets(data, nb_packets); + * if (out_len < 0) + * return (int)out_len; + * output_next_packet(out, out_len); + * } + * @endcode + * + * An alternate way of merging packets is to simply call opus_repacketizer_cat() + * unconditionally until it fails. At that point, the merged packet can be + * obtained with opus_repacketizer_out() and the input packet for which + * opus_repacketizer_cat() needs to be re-added to a newly reinitialized + * repacketizer state. + */ + +typedef struct OpusRepacketizer OpusRepacketizer; + +/** Gets the size of an <code>OpusRepacketizer</code> structure. + * @returns The size in bytes. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_size(void); + +/** (Re)initializes a previously allocated repacketizer state. + * The state must be at least the size returned by opus_repacketizer_get_size(). + * This can be used for applications which use their own allocator instead of + * malloc(). + * It must also be called to reset the queue of packets waiting to be + * repacketized, which is necessary if the maximum packet duration of 120 ms + * is reached or if you wish to submit packets with a different Opus + * configuration (coding mode, audio bandwidth, frame size, or channel count). + * Failure to do so will prevent a new packet from being added with + * opus_repacketizer_cat(). + * @see opus_repacketizer_create + * @see opus_repacketizer_get_size + * @see opus_repacketizer_cat + * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to + * (re)initialize. + * @returns A pointer to the same repacketizer state that was passed in. + */ +OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1); + +/** Allocates memory and initializes the new repacketizer with + * opus_repacketizer_init(). + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusRepacketizer *opus_repacketizer_create(void); + +/** Frees an <code>OpusRepacketizer</code> allocated by + * opus_repacketizer_create(). + * @param[in] rp <tt>OpusRepacketizer*</tt>: State to be freed. + */ +OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp); + +/** Add a packet to the current repacketizer state. + * This packet must match the configuration of any packets already submitted + * for repacketization since the last call to opus_repacketizer_init(). + * This means that it must have the same coding mode, audio bandwidth, frame + * size, and channel count. + * This can be checked in advance by examining the top 6 bits of the first + * byte of the packet, and ensuring they match the top 6 bits of the first + * byte of any previously submitted packet. + * The total duration of audio in the repacketizer state also must not exceed + * 120 ms, the maximum duration of a single packet, after adding this packet. + * + * The contents of the current repacketizer state can be extracted into new + * packets using opus_repacketizer_out() or opus_repacketizer_out_range(). + * + * In order to add a packet with a different configuration or to add more + * audio beyond 120 ms, you must clear the repacketizer state by calling + * opus_repacketizer_init(). + * If a packet is too large to add to the current repacketizer state, no part + * of it is added, even if it contains multiple frames, some of which might + * fit. + * If you wish to be able to add parts of such packets, you should first use + * another repacketizer to split the packet into pieces and add them + * individually. + * @see opus_repacketizer_out_range + * @see opus_repacketizer_out + * @see opus_repacketizer_init + * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to which to + * add the packet. + * @param[in] data <tt>const unsigned char*</tt>: The packet data. + * The application must ensure + * this pointer remains valid + * until the next call to + * opus_repacketizer_init() or + * opus_repacketizer_destroy(). + * @param len <tt>opus_int32</tt>: The number of bytes in the packet data. + * @returns An error code indicating whether or not the operation succeeded. + * @retval #OPUS_OK The packet's contents have been added to the repacketizer + * state. + * @retval #OPUS_INVALID_PACKET The packet did not have a valid TOC sequence, + * the packet's TOC sequence was not compatible + * with previously submitted packets (because + * the coding mode, audio bandwidth, frame size, + * or channel count did not match), or adding + * this packet would increase the total amount of + * audio stored in the repacketizer state to more + * than 120 ms. + */ +OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2); + + +/** Construct a new packet from data previously submitted to the repacketizer + * state via opus_repacketizer_cat(). + * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to + * construct the new packet. + * @param begin <tt>int</tt>: The index of the first frame in the current + * repacketizer state to include in the output. + * @param end <tt>int</tt>: One past the index of the last frame in the + * current repacketizer state to include in the + * output. + * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to + * store the output packet. + * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in + * the output buffer. In order to guarantee + * success, this should be at least + * <code>1276</code> for a single frame, + * or for multiple frames, + * <code>1277*(end-begin)</code>. + * However, <code>1*(end-begin)</code> plus + * the size of all packet data submitted to + * the repacketizer since the last call to + * opus_repacketizer_init() or + * opus_repacketizer_create() is also + * sufficient, and possibly much smaller. + * @returns The total size of the output packet on success, or an error code + * on failure. + * @retval #OPUS_BAD_ARG <code>[begin,end)</code> was an invalid range of + * frames (begin < 0, begin >= end, or end > + * opus_repacketizer_get_nb_frames()). + * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the + * complete output packet. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Return the total number of frames contained in packet data submitted to + * the repacketizer state so far via opus_repacketizer_cat() since the last + * call to opus_repacketizer_init() or opus_repacketizer_create(). + * This defines the valid range of packets that can be extracted with + * opus_repacketizer_out_range() or opus_repacketizer_out(). + * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state containing the + * frames. + * @returns The total number of frames contained in the packet data submitted + * to the repacketizer state. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1); + +/** Construct a new packet from data previously submitted to the repacketizer + * state via opus_repacketizer_cat(). + * This is a convenience routine that returns all the data submitted so far + * in a single packet. + * It is equivalent to calling + * @code + * opus_repacketizer_out_range(rp, 0, opus_repacketizer_get_nb_frames(rp), + * data, maxlen) + * @endcode + * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to + * construct the new packet. + * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to + * store the output packet. + * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in + * the output buffer. In order to guarantee + * success, this should be at least + * <code>1277*opus_repacketizer_get_nb_frames(rp)</code>. + * However, + * <code>1*opus_repacketizer_get_nb_frames(rp)</code> + * plus the size of all packet data + * submitted to the repacketizer since the + * last call to opus_repacketizer_init() or + * opus_repacketizer_create() is also + * sufficient, and possibly much smaller. + * @returns The total size of the output packet on success, or an error code + * on failure. + * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the + * complete output packet. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1); + +/** Pads a given Opus packet to a larger size (possibly changing the TOC sequence). + * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the + * packet to pad. + * @param len <tt>opus_int32</tt>: The size of the packet. + * This must be at least 1. + * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding. + * This must be at least as large as len. + * @returns an error code + * @retval #OPUS_OK \a on success. + * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len. + * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet. + */ +OPUS_EXPORT int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len); + +/** Remove all padding from a given Opus packet and rewrite the TOC sequence to + * minimize space usage. + * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the + * packet to strip. + * @param len <tt>opus_int32</tt>: The size of the packet. + * This must be at least 1. + * @returns The new size of the output packet on success, or an error code + * on failure. + * @retval #OPUS_BAD_ARG \a len was less than 1. + * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_packet_unpad(unsigned char *data, opus_int32 len); + +/** Pads a given Opus multi-stream packet to a larger size (possibly changing the TOC sequence). + * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the + * packet to pad. + * @param len <tt>opus_int32</tt>: The size of the packet. + * This must be at least 1. + * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding. + * This must be at least 1. + * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet. + * This must be at least as large as len. + * @returns an error code + * @retval #OPUS_OK \a on success. + * @retval #OPUS_BAD_ARG \a len was less than 1. + * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet. + */ +OPUS_EXPORT int opus_multistream_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len, int nb_streams); + +/** Remove all padding from a given Opus multi-stream packet and rewrite the TOC sequence to + * minimize space usage. + * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the + * packet to strip. + * @param len <tt>opus_int32</tt>: The size of the packet. + * This must be at least 1. + * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet. + * This must be at least 1. + * @returns The new size of the output packet on success, or an error code + * on failure. + * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len. + * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet. + */ +OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_packet_unpad(unsigned char *data, opus_int32 len, int nb_streams); + +/**@}*/ + +#ifdef __cplusplus +} +#endif + +#endif /* OPUS_H */ |